The present invention relates to an apparatus for signal processing discrete-time values for signal-sampling systems having means, such as digital/analog converters, switched capacitor filters, direct digital synthesizers, sample-and-hold circuits and the like, which deliver an analog discrete-time output signal, having means for producing discrete-time values, which are supplied to the means delivering an analog discrete-time output signal, at least one group of at least two parallel-connected signal-sampling means, means for producing sampling signals at the same frequency, summation means which sum the output signals from the signal-sampling means.
It is known that sampling systems can output particular values only at particular times. This means that the output signal for such a system is similar to a staircase and thus also contains spectral components of the sampling frequency in addition to the useful information. These spectral components are suppressed by a time-continuous filter so that no convolution products are produced from the useful signal and the sampling frequency.
Problems arise in known digital/analog (DA) converters. Such problems are easily explained using a compact disk (CD) player. The simplified block diagram of the reproduction electronics of a first generation CD player is illustrated in FIG. 2. The reproduction electronics 50 deliver a digital data stream at 16 bits per clock pulse. The sampling frequency is 44.1 kHz. Since the audio band ranges from 20 Hz to 20 kHz, it is possible, in accordance with the sampling theorem, to restore the original information using a digital/analog converter 52 clocked at 44.1 kHz and using a downstream analog anti-aliasing filter 54 which suppresses convolution products. This filter 54 has two fundamental requirements to fulfil. It needs to have a smooth frequency response in the useful range and needs to have an attenuation of more than 60 dB at half the sampling rate of 22.05 kHz, in order to suppress undesirable aliasing products. Such requirements are met only using complicated 13th order filters 54.
Another solution came to light with the improvement of DA converter technology. The second generation used a 2-fold to 4-fold oversampling rate. FIG. 3 is a schematic illustration of a block diagram of the reproduction electronics of second generation CD players. For this, special digital filters inside the reproduction electronics 56xe2x80x94so-called FIR filters (Finite Impulse Response Filter)xe2x80x94were used to calculate intermediate values, which were additionally output to the DA converter 58. This increased the sampling rate from 44.1 kHz to 88.2 kHz or 176.4 kHz. However, in addition to the increased sampling frequency, the DA converter 58 also needed to be even more accurate, namely 20 bits. The requirements to be more accurate and faster actually conflict with one another. The advantage obtained at the price of this cost-intensive solution is that the analog filter is simplified because the 60 dB attenuation need be achieved only at half the clock rate 44.1 kHz or 88.2 kHz. With 4-fold oversampling, the number of necessary poles in the filter 60 is reduced to 5. In the case of 16-fold oversampling, which is possible today, second order filters are in fact adequate.
Switched capacitor filters are likewise sampling systems, which means that the same problems arise in this case. Active time-continuous filters require resistors and capacitors in addition to the active element (e.g. operational amplifiers). In switched capacitor filters, the resistors are replaced by switched capacitors. There is a linear relationship between the switching frequency and the equivalent admittance. Typical sampling frequencies of integrated SC filters are 50 to 100 times the cutoff frequency. This means that the output signal is composed of 50 to 100 analog discrete-time values. This staircase-like profile thus also contains spectral components of the switching frequency. These can be suppressed by a filter downstream. If this filter is intended to be made tunable by varying the sampling frequency, a range of two decades can thus be implemented. If, by way of example, it is desirable to produce a low-pass filter covering the audio range, then the range to be tuned extends over three decades from 20 Hz to 20 kHz. This would mean that the analog smoothing filter would likewise have to be designed so that it can be tuned or switched over, because, if this filter has a cutoff frequency of 20 Hz, it generates a frequency component, depending on the system, of 2 kHz with an amplitude of around xe2x88x9240 dB (1/100) below the useful signal.
Further prior art relating to DA converters and SC filters can be found in the following publications:
Paul Horowitz, The Art of Electronics, Cambridge University Press, Cambridge 1990,
Robert Adams, DAC ICs: How Many Bits is Enough?, Sound and Video Contractor, Feb. 20, 1991, pages 8-189 to 8-192,
Larry Gaddy and Hajima Kawai, DYNAMIC PERFORMANCE TESTING OF DIGITAL AUDIO D/A CONVERTERS, APPLICATION BULLETIN, Burr-Brown Corporation, 1997,
Helmuth Lemme, Filtern ohne zu rechnen [Filtering without calculation], Elektronik November 1997, pages 96 to 104,
Internet address: http://www.km.philips.com/osc/cdrom/geninfo/geninfxe2x80x941.html, Nov. 4, 1997, pages 1 to 18,
Nav S. Sooch et al., 18-BIT STEREO D/A CONVERTER WITH INTEGRATED DIGITAL AND ANALOG FILTERS, Sooch CS4328 AES Paper, New York, October 1991,
Digital-to-Analog Converter with Low Intersample Transistion [sic] Distortion and Low Sensitivity to Sample Jitter and Transresistance Amplifier Slew Rate, in Audio Engineering Society, Vol. 42, No. 11, 1994 November,
Section 6.11 Glitchless DACs, in the book Analog-to-Digital and Digital-to-Analog Conversion Techniques, by David F. Hoeschele Second Edition Wiley-Interscience 1994 ISBN 0-471-57147-4.
U.S. Pat. No. 5,521,946 discloses an apparatus for signal processing discrete-time values of the type mentioned in the introduction. In this apparatus, a plurality of DA converters are connected in parallel and have different values applied to them. The same input signals are supplied to different digital filters which, in turn, produce different digital output values representing the different input signals for the DA converters. After conversion into analog signals, summation is carried out in the analog domain.
The present invention is based on the object or the technical problem of specifying, on the basis of the aforementioned prior art, an apparatus for signal processing discrete-time values of the type mentioned in the introduction which enables the number of intermediate values in sampling systems to be increased without altering the accuracy or the sampling rate of the sampling system.
The apparatus according to the present invention generally comprises an apparatus for signal processing discrete-time values for signal-sampling systems having means, such as digital/analog converters, switched capacitor filters, direct digital synthesizers, sample-and-hold circuits and the like, which deliver, an analog discrete-time output signal, having means for producing discrete-time values, which are supplied to the means delivering an analog discrete-time output signal. The apparatus also includes at least one group of at least two parallel-connected signal-sampling means and also means for producing sampling signals at the same frequency. In addition, summation means are provided which sum the output signals from the signal-sampling means.
Accordingly, the apparatus according to the invention for signal processing discrete-time values is distinguished in that, in the at least one group of at least two signal-sampling parallel-connected means, signal sampling is in each case carried out on the input signal, which is identical in terms of magnitude, and the respective sampling means is driven by the means for producing sampling signals at the same frequency using a shifted phase angle for the sampling signals, the result of which is that, for each input signal, a plurality of analog output signals are produced which are identical but phase-shifted on the basis of the number of parallel-connected signal-sampling means, the summation of said output signals representing interpolation in the analog domain.
One particularly preferred refinement is distinguished in that the number of parallel-connected sampling means is equivalent to the number of frequencies produced with a shifted phase angle.
One particularly preferred refinement of the apparatus according to the invention is distinguished in that the phase differences of the sampling frequencies with a shifted phase angle are identical, that is to say that n sampling signals each phase-shifted through 360xc2x0/n are produced, where n corresponds to the number of parallel-connected sampling means, that is to say that linear interpolation is possible.
One alternative refinement is distinguished in that, in the case of sampling signals at the same frequency, the phase differences of the sampling frequencies with a shifted phase angle are different. This enables nonlinear interpolation to be implemented, which makes it possible, by way of example, for ranges with a particularly high level of accuracy or other, for example singular, ranges to be reliably depicted.
One advantageous development is characterized in that the means sampling with a different phase angle are split into two groups which have the uninverted input signal and the inverted input signal applied to them, respectively, and the summing means sums the uninverted signals and deducts the inverted signals.
One particularly preferred development is distinguished in that at least two sampling means are identical.
One particularly advantageous refinement is distinguished in that a plurality of groups of systems which are to be sampled are provided, each group being driven using different sampling frequencies with a shifted phase angle. This means that, as an alternative, the further processing of the signals can preferably be devised such that the signals from the plurality of groups of sampling means are supplied to a summing means, or can preferably be devised such that the summed signals from the first group of sampling means are supplied to a second group of sampling means, which, in turn, are driven using sampling frequencies with a shifted phase angle and whose signals are supplied to further summing means.
A further preferred refinement of the apparatus according to the invention is distinguished in that a plurality of groups of means which are to be sampled are provided, each group being driven using identical sampling frequencies with a shifted phase angle.
Further embodiments and advantages of the invention are revealed by the additional features set out in the claims and by the illustrative embodiments specified below. The features of the claims can be combined with one another in any desired manner, provided that they are not obviously mutually exclusive.